Implements peer connection stats measurements in WebRTC call test.

tools/perf/page_sets/webrtc_cases.py

 # Copyright 2014 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
+import os
+
 from telemetry.page import page as page_module
 from telemetry.page import page_set as page_set_module
 
 
 class WebrtcCasesPage(page_module.Page):
 
   def __init__(self, url, page_set):
     super(WebrtcCasesPage, self).__init__(url=url, page_set=page_set)
 
+    with open(os.path.join(os.path.dirname(__file__),
+                           'webrtc_track_peerconnections.js')) as javascript:
+      self.script_to_evaluate_on_commit = javascript.read()
+
 
 class Page1(WebrtcCasesPage):
 
   """ Why: Acquires a vga local stream. """
 
   def __init__(self, page_set):
     super(Page1, self).__init__(
       url=('http://googlechrome.github.io/webrtc/samples/web/content/'
            'getusermedia-resolution/'),
       page_set=page_set)
@@ skipping 40 matching lines @@
   """ WebRTC tests for Real-time audio and video communication. """
 
   def __init__(self):
     super(WebrtcCasesPageSet, self).__init__(
       archive_data_file='data/webrtc_cases.json',
       bucket=page_set_module.PUBLIC_BUCKET)
 
     self.AddPage(Page1(self))
     self.AddPage(Page2(self))
     self.AddPage(Page3(self))